The transmission of low bit rate voice utilizing packets while offering the prospects of significant savings in transmission and switching costs over circuit switching has not been widely used because of problems encountered in transmitting encoded voice information via packets. One of these problems has been the variable delay experienced by the packets during transmission through a packet switching system. The delay variation is introduced in the switching nodes of the system and results from varying traffic load conditions; whereas, the delay occurring on the transmission links interconnecting the nodes is constant. Unlike data, voice information must be decoded from the digital representation to the analog representation at a fixed rate in order to maintain quality voice reproduction. The introduction of a variable delay during the transmission of the voice packets does not allow this uniform decoding to take place unless some compensation is made for the variable delay. The compensation performed depends upon whether the packet arrived before or after the time for the decoding of the packet. If the packet arrives early, then the decoding is delayed for the proper amount of time. If the packet arrives late, when it is necessary to interpolate during the decoding operation to obtain correct analog signals. But before such compensation can be performed, the delay must be accurately measured.
One solution in prior art systems for the switching of voice packets has resolved this problem by including in each packet a time stamp designating when the packet was transmitted from the originating customer terminal. When the packet arrives at the destination customer terminal, the delay encountered by the packet is determined by comparing the time in the time-stamp field with the present time. The problem encountered with this prior art solution is that all of the customer terminals connected to the packet switching system must maintain synchronized clocks. This solution was acceptable in prior art systems which consisted of a few customer terminals. However, in a large packet switching system having thousands of customer terminals, the prior art solution is not a practical solution, since there are too many technical difficulties in maintaining synchronized clocks over a large number of customer terminals.
A second prior art solution took advantage of the fact that low bit rate encoded voice occurs in bursts of information. This fact results in a number of packets being transmitted during the active portion of the conversation followed by no packets being transmitted during the silent portion. The prior art solution was to delay the first packet of an active portion of the conversation by the maximum amount of time which a packet could be delayed during transmission through the packet switching system. After this initial delay, all other received packets were decoded at a fixed rate. The disadvantage of this method was that the reproduction of the analog voice signals could be delayed by an unacceptable amount since the reproduction is delayed by maximum transmission plus the time required for the actual transmission of the first packet. This was particularly true if the transmission of voice was over large distances.
In view of the foregoing, there exists a need for improved techniques and facilities for accurately determining the delay encountered by a packet as it progresses through a packet switching system without adding further delay to the packet transmission or requiring synchronization of the clocks in customer terminals. Such techniques and facilities should allow uniform and accurate compensation of voice packets by providing accurate information with respect to the delay encountered by packets during transmission.